In today's technological environment, many ways exist for several people in multiple geographic locations to communicate with one another simultaneously. One such way is audio conferencing. Audio conferencing applications serve both the needs of business users and leisure users who are geographically distributed.
Traditional audio conferencing involved a central conferencing server which hosted an audio conference. Participants used their telephones to dial in to the conferencing server over the Public Service Telephone Network (PSTN) (also called the Plain Old Telephone System (POTS)).
Greater availability of low-cost personal computers, networking equipment, telecommunications, and related technology, however, has dramatically changed the way people communicate. One example of such change is the tremendous increase in persons connected to the global Internet. Connectivity achieved by the Internet—connecting numerous, different types of networks—is based upon a common protocol suite utilized by those computers connecting to it. Part of the common protocol suite is the Internet Protocol (IP), defined in Internet Standard (STD) 5, Request for Comments (RFC) 791 (Internet Architecture Board). IP is a network-level, packet (i.e., a unit of transmitted data) switching protocol.
In recent years, technological improvements offer the possibility of transmitting voice data over the worldwide public Internet. Voice over IP (VoIP) began with computer scientists experimenting with exchanging voice using personal computers (PCs) equipped with microphones, speakers, and sound cards.
VoIP further developed when, in March of 1996, the International Telecommunications Union-Telecommunications sector (ITU-T), a United Nations organization, adopted the H.323 Internet Telephony Standard. Among its specifications, H.323 provides the minimum standards that equipment must meet in order to send voice over the IP, and other packet-switched network protocols where quality of sound cannot be guaranteed. Thus, conferencing servers (also called multipoint control units (MCUs)) were developed to host audio conferences where participants connected to a central MCU using PC-based equipment and the Internet, rather than traditional phone equipment.
More recently, several alternatives to H.323 have been developed. One such alternative is the Session Initiation Protocol (SIP) developed within the Internet Engineering Task Force (IETF) Multiparty Multimedia Session Control (MMUSIC) Working Group. SIP, which is well-known in the relevant art(s), is a signaling protocol for Internet conferencing and telephony. SIP addresses users using an e-mail-like address and utilizes a portion of the infrastructure used for Internet e-mail delivery. It handles basic setup functions as well as enhanced services (e.g., call forwarding).
Given the rapid pace of development in the telephony industry—both in protocols and equipment—and the existence of both legacy equipment and protocols (e.g., telephones and switching networks such as the PSTN), audio conferencing service providers need a means to link legacy circuit-switched systems to newer packet-switched systems in order to reach (or service) a broader range of clients and vice versa. Therefore, a method is needed to seamlessly link a combination of MCU architectures for packet based (e.g., IP-based) client and circuit switched (e.g., phone) based client conferencing. The linkage of this combination of MCUs should realize the capabilities of the various participants' equipment and provide the appropriate audio data to each participant.